Hi all,
Does anybody know a free test-tool to test the performance of e.g. a
DSL-modem regarding the short packets used by G.729?
I recently used iperf, but this tool produces bursts and does not simulate
the steady stream of media packets very well.
Regards
Franz
Hi all,
I was exploring the possibility for the UPDATE to send the Answer,
As I haven't found any explicit restriction in RFC 3311.
I went through the older Thread
http://lists.cs.columbia.edu/pipermail/sip-implementors/2004-January/006017.
html
Hi
No UPDATE cannot be used to send answer, since UPDATE can be used only
after a dialog is established.
Regards
Ranjit
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Chauhan Bhavik-A20762
Sent: Thursday, May 19, 2005 3:10 PM
To:
Hi
I have a query regarding PTT
If simultaneously 10 users press the button
who will get the priority and on what basis
OR NO ONE WILL GET PRIORITY
VISHAL
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Sip-implementors mailing list
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Hi,
I just thought I'd try to clarify something:
3. Is 18x send/recv over TCP considered reliable ?
In the following scenario:
- INVITE with offer sdp sent
- 18x over TCP with answer sdp recvd (No PRACK exchange happened)
Now can the caller/calleee use UPDATE ?
Can any body
Hi,
If you don't use PRACK you can't send a valid offer in a 18x. An offer (and an
answer) has to be sent reliably.
Regards,
Christer Holmberg
LM Ericsson
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chauhan
Bhavik-A20762
Sent: 19. toukokuuta
Hello all,
RFC 3261 says Before a request is sent, the client transport MUST insert a
value of the sent-by field with FQDN or IP and Port into the Via header field.
I dont see an UAC inserting a sent-by in the VIA header.Normally a VIA header
would look like this,
INVITE sip:[EMAIL
We really need to create an FAQ about this.
A long time ago, probably in the thread you cite, we debated this
extensively. It is true that the RFCs are not entirely clear on this,
and reasonable people can differ in interpretation of them. But we
finally came to some agreement on it.
I am
Hi,
Actaully in my application there is no fecility to
domain name mapping,i mean it is having an option for
only IP it is not having fecility for DNS(i mean for
interop.pingtel.com).that's why i'm unable to connect
ur's online servers.
please give me solution for this,
Thanks and regards,
From: Todd Huang
TransferorTransferee Transfer
|| Target
|||
dialog1 | INVITE/200 OK/ACK |
From: Anil Bollineni [EMAIL PROTECTED]
Assume a gateway sends two INVITEs say to two different UASs with the
same
Via branch parameter to two different users at the same time.
It must not do that. See section 8.1.1.7 of RFC 3261: The branch
parameter value MUST be unique across space and
From: Anil Bollineni [EMAIL PROTECTED]
I want to talk about NAT firewall. If two UA's behind NAT firewall
generate same
branch, and NAT ALG will change the VIA sent-by field to same value. In
this
case it could result matching same transaction.
True. Which is why UAs should generate branch
Dear Sirs,
I am a newbie and please forgive me if this post does not below in this list.
I have a question that I hope you might be able to clarify for me. Gateway A
sends an INVITE to Gateway B with SDP. When B sends back 183 Session Progress
with SDP, shouldn't A respond and use the
It depends upon what is carried in the 183 SDP.
Let us say 183 Is carrying a SDP which connects A to a Media Server and
Media Server is just playing an announcement, that your call is proceeding.
In that case you would not want to start billing that person after receiving
media in 183.
200 OK
Hi,
Thank you Indresh for your response. I agree with you that we should not
be billing early until a connection has been established. During this call,
the billing did not start until we (10.1.26.125) sent 200 OK SDP. The thing
I would like to understand is why does it take like 23seconds
That is not PDD. PDD is time from the iNVITE to any of the following:
CANCEL
180
183
2xx
3xx
4xx
5xx
Response. So the time between the invite and the 183 is your PDD.
The 23 seconds you mention, is how long the phone rangor the time between
your 183 and the 200 is your ring time.
Pong,
I was going to ask you why the long delay between the 183 and 200 !,
I guess you beat me to it.
The trace is helpful but doesn't explain the delay. The field Resent
Packet: False might not mean much depending on where the trace was taken
and over what transport protocol the
Does this Proxy server support Presence?
Rgds
jaikumar
- Original Message -
From: Scott Lawrence [EMAIL PROTECTED]
To: sip-implementors@cs.columbia.edu
Sent: Wednesday, May 18, 2005 11:49 PM
Subject: [Sip-implementors] Pingtel Test Proxy Available
Pingtel is making available a public
Yes Agree, The FAQ creation afford would be appreciated and this will reduce
exploration through threads which can be like debate than conclusion and
in fact this is quite natural.
Also I would be thankful if we maintain agreed upon things which are not
cited in RFCs
to ascertain the unanimity
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